1. Field of the Invention
This invention relates to computer networks and, more specifically to the reservation of bandwidth in computer networks.
2. Background Information
Computer networks typically comprise a plurality of interconnected entities. An entity may consist of any device, such as a computer or end station, that “sources” (i.e., transmits) or “sinks” (i.e., receives) datagrams (e.g., packets and/or frames). A common type of computer network is a local area network (“LAN”) which typically refers to a privately owned network within a single building or campus. LANs typically employ a data communication protocol (LAN standard), such as Ethernet, FDDI or token ring, that defines the functions performed by the data link and physical layers of a communications architecture (i.e., a protocol stack). In many instances, several LANs may be interconnected by point-to-point links, microwave transceivers, satellite hook-ups, etc. to form a wide area network (“WAN”) or intranet that may span an entire country or continent.
One or more intermediate network devices are often used to couple LANs together and allow the corresponding entities to exchange information. For example, a bridge may be used to provide a “bridging” function between two or more LANs. Alternatively, a switch may be utilized to provide a “switching” function for transferring information between a plurality of LANs or end stations. Bridges and switches may operate at various levels of the communication protocol stack. For example, a switch may operate at layer 2 which, in the Open Systems Interconnection (OSI) Reference Model, is called the data link layer and includes the Logical Link Control (LLC) and Media Access Control (MAC) sub-layers. Data frames at the data link layer typically include a header containing the MAC address of the entity sourcing the message, referred to as the source address, and the MAC address of the entity to whom the message is being sent, referred to as the destination address. To perform the switching function, layer 2 switches examine the MAC destination address of each data frame received on a source port. The frame is then switched onto the destination port(s) associated with that MAC destination address.
Other network devices, commonly referred to as routers, may operate at higher communication layers, such as layers 3 and 4 of the OSI Reference Model, which in Transmission Control Protocol/Internet Protocol (TCP/IP) networks corresponds to the IP and the TCP/User Datagram Protocol (TCP/UDP) layers. Data frames at the IP layer include a header which contains an IP source address and an IP destination address, while frames at the TCP/IP layer include source and destination port numbers. Routers or layer 3 switches may re-assemble or convert received data frames from one LAN standard (e.g., Ethernet) to another (e.g. token ring). Thus, layer 3 devices are often used to interconnect dissimilar subnetworks.
Voice Over IP (VoIP)
Traditionally, computer networks were used to exchange static files or data, such as text and spreadsheet files, while the Public Switched Telephone Network (PSTN) was used to exchange voice information. Computer networks, however, are increasingly being used to transport “voice” information. Voice over IP (VoIP) typically refers to a group of technologies used to transmit voice information over computer networks. Such networks include a plurality of voice agents that convert voice information from its traditional telephony form to a form that is suitable for packet transmission. In other words, the voice agent encodes, compresses and encapsulates the voice information into a plurality of data packets. Examples of voice agents include IP telephones, VoIP gateways, certain private branch exchanges (PBXs), personal computers (PCs) running communication applications, network devices providing voice gateway services, etc. A calling party uses a voice agent to initiate a VoIP call. Once the voice information has been converted into packet format, it is carried by the computer network to a second voice agent configured to serve the called party. Voice traffic, unlike static data files or records, is highly sensitive to delay and to lost packets. That is, delays in receiving data packets carrying voice information at the called party's voice agent can seriously degrade the quality of the call. Accordingly, packets carrying voice information must be delivered to the called party with high probability and in a timely manner.
Computer networks include numerous services and resources for use in forwarding network traffic. For example, different network links, such as Fast Ethernet, Asynchronous Transfer Mode (ATM) channels, SONET links, satellite links, etc., offer different speed and bandwidth capabilities. Particular intermediate devices also include specific resources or services, such as priority queues, filter settings, traffic shapers, queue selection strategies, congestion control algorithms, etc. that affect the rate at which traffic moves through the device and thus across the network. Depending on the selection or allocation of such resources or services, network traffic for different sources and sinks can be forwarded at different speeds or rates, thereby controlling the loss and/or delay experienced by the traffic.
The Resource Reservation Protocol
As set forth above, to support VoIP, packets carrying voice information must typically be delivered within narrow time constraints. Although many computer networks have the resources and services to meet the delivery requirements of VoIP, these resources and services must be allocated, preferably in advance, to the correct network traffic. The Resource reSerVation Protocol (RSVP), which is set forth at RFC 2205, is a signaling protocol that was developed so that entities (typically referred to as receivers) could reserve bandwidth within their computer networks to receive from one or more sourcing entities a desired traffic flow, such as multimedia stream. Pursuant to RSVP, sources send RSVP Path messages identifying themselves and indicating the bandwidth needed to receive their programming or content. These messages proceed hop-by-hop through the intermediate network devices, making those devices aware of the possibility that a reservation of resources may be required. If a receiver is interested in the programming or content offered by a particular source, it responds with a RSVP Reservation (Resv) message, which travels hop-by-hop back to the source. At each hop, the corresponding intermediate device establishes a session for the receiver and sets aside sufficient resources to provide the requested bandwidth for the desired traffic flow. These resources are immediately made available to the traffic flow. If the resources are not available, the reservation is refused explicitly so that the receiver knows it cannot depend on the corresponding resources being devoted to its traffic. By using RSVP, packets carrying voice information can be accorded the resources and services they need to ensure timely delivery.
VoIP telephones and voice applications are increasingly being offered with greater feature sets. One such feature is “call waiting”, which is a common service or feature of conventional, analog telephone sets. With call waiting, a first party who is on the phone with a second party is alerted to an incoming call from a third party. The first party can put the second party on hold and answer the call from the third party. The first party can even alternate back and forth between second and third parties. VoIP telephones and voice applications support of call waiting, however, can create potential problems, such as the setting aside of resources unnecessarily.
In particular, the VoIP telephone or application will typically use RSVP to reserve separate resources along the entire route to both the second and third parties to support the two calls. However, because call waiting only allows the first party to talk to either the second or the third party at any given point in time, only one set of the resources will be in use at any instant. That is, at any given time, the VoIP telephone or application will be generating packets for forwarding to either the second party or the third party, but not both. Nonetheless, resources will remain reserved to support traffic to both the second and third parties at all times. As a result, resources that could be applied to other traffic are wasted. In addition, the request to reserve resources for the third party may fail admission control at one or more devices along that portion of the route shared with the second party even though a separate set of resources are really not required. This would result in the voice traffic to or from the third party being denied access to adequate resources and thus severely degrading the “quality” of the call, even though resources set aside for the second party remain idle.